similar to: SIP-Channels cannot be created after a while of running asterisk ...

Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following

Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I

Hi all, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice

Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called

Hello everybody! After checking the complete wiki and the mailinglist archives I still haven't really found out why the following constellation does not work. We have an asterisk-System with some SIP-Phones and an old ISA Fritz-ISDN-Card used with i4l. The whole system is integrated in out (ISDN-)PBX for testing. The ISDN-Card is properly configured, as we are able to phone out, receive

Hi folks! Is it possible to tell asterisk not to strip the leading 0 of *incoming* MSNs? I use asterisk with i4l and whenever I get a call from an long-distance party, the leading 0, which should be there according the german numbering, is not. So if I get a call from a mobile phone 0177-1234567 should be displayed, but 177-1234567 is displayed. I double checked if I've forgotten to remove an

Hi Well, subject says all. In Chapter 9.2.2.1, TBF, the parameter mpu or "minimum packet size" is explained as: > A zero-sized packet does not use zero bandwidth. For ethernet, no packet > uses less than 64 bytes. The Minimum Packet Unit determines the minimal > token usage for a packet. In my understanding an ethernet packet needs at least 14 (2*6+2) bytes or 54 bytes if

> Is it possible to tell asterisk not to strip the leading 0 > of *incoming* MSNs? I use asterisk with i4l and whenever > I get a call from an long-distance party, the leading 0, which > should be there according the german numbering, is not. Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped already by the phone company? I think the easiest

This is super easy to do. All you need to do is to put that announcement in a MP3 and set a musiconhold class for that incoming Zap channel. Then basically when ever that PSTN number rings, Asterisk will play the MP3 stream "Your call may be monitored or recorded, please hangup if you do not agree...etc" in a loop until the line is answered. Caller doesn't pay a single dime to

this : iptables -A FORWARD -i internal-interface -j ACCEPT iptables -A FORWARD -m state --state ESTABLISHED,RELATED -j ACCEPT iptables -A FORWARD -j DROP doesn''t seem to work for active-ftp .. i even manualy loaded ip_conntrack_ftp but as u see it is unused : # lsmod Module Size Used by Not tainted ip_conntrack_ftp 4272 0 (unused) iptable_nat

Dear R-users, I have a set of 12000 image samples. I can divide this set into two categories: training and testing. I need to classify the test set into a two qualitative outputs: true or false for some characteristic. To do the classification I'm using the packages SVM in e1071 library and LDA in the MASS library. However, I'm with a great number of FALSE POSITIVE CASES in both

I have a Mandrake 10.0 Official server running Samba3, Shorewall, Squid 2.5, Postfix and Courier-IMAP. Samba uses winbind to authenticate mail and proxy users against a windows 2000 ADS server. I get corruption happening in the user's home directories and elsewhere. The directory ownership changes all the time. One moment a dir belongs to roland:Domain Users and the next moment it's

Hi guys, I am starting project to develop open implementation of some (naturally open codec) for simply designed embedded devices which can be used by small to big VoIP operators or hobbyists which wants to build their own small and low cost VoIP phone. Speex seems to be great choice. And greater will be if there should be some guys who want to help ;-). Altought I am experienced in C/ASM

Hi I am facing a problem where I would like to import a TIFF image (of spots on a nylon filter) into R (into a matrix for example). When plotting the matrix using fx. scatterplot3d I would then be able to see how the pixel-intensities are distributed in "spot-areas" on the filter - which would be very helpful. Does anynone know of a way to do this? Best regards, Peter Hagedorn ...

Hi All, I have been researching Asterisk for a few days now and have read hundreds of web pages and other documents. While some things are getting clearer, others are not. I have managed to install Asterisk 1.0.1 on Debian testing (simple as 'apt-get install asterisk'). I understand FXS/FXO, codecs available, feature and functionality available etc. I have found several "Getting

When I try to push my app to Heroku I get this error: OCA@OCA-PC /c/rails/oca (master) $ git push heroku master Enter passphrase for key ''/c/Users/OCA/.ssh/id_rsa'': Counting objects: 913, done. Delta compression using up to 2 threads. Compressing objects: 100% (797/797), done. Read from remote host heroku.com: Connection reset by peer fatal: sha1 file

Rádio IRCBrasil - André Marcelo wrote: > Hi people, > > Tody i am changing my transmition system from shoutcast to icecast 2, > encoding in ogg > > > My listenners are telling me that they cannot listen with winamp 2.91 lite > and winam3, just winamp 2.91 standard or full That is correct. Winamp 3 is broken, and will not correctly play Ogg streams. Winamp 2.91 lite is

Hi people, Tody i am changing my transmition system from shoutcast to icecast 2, encoding in ogg <p>My listenners are telling me that they cannot listen with winamp 2.91 lite and winam3, just winamp 2.91 standard or full Just anynone know how to listen with winamp 3 ? listen my radio in http://play.ircbrasil.com.br in browser or www.ircbrasil.com.br:8500/ircbrasil.ogg in winamp tanks 4

Hi,When I read the source of str,i find these code ----- ## Show further classes // Assume that they do NOT have an own Method -- ## not quite perfect ! (.Class = 'remaining classes', starting with current) cl <- oldClass(object); cl <- cl[cl != "data.frame"] #- not THIS class ----- so I use ?oldClass to try to learn more about oldClass.But after I have reading

I have a couple development servers running centos 6.3 64bit that have LSI 9211-8i SAS2 controllers connected to a SAS2 backplane. these work fine with SATA hard disks (populated with a bunch of 3TB SATA drives)... I'm trying to install a OCZ Vertex3 SSD on each of the two servers to do some ssd caching tests... system sees the drive, so I do the following... # parted -a min